Audio level meter

ABSTRACT

A circuit for correcting the output of an audio level meter comprises input means for generating the square or the absolute value of an input signal, a low pass filter having a predetermined attack time and release time, and output means for converting the output signal from a linear scale to a logarithmic scale. The circuit further comprises a correction means to which an information about whether the input signal to the audio level meter was subject to squaring or converting into an absolute value at the input, as well as the attack and release time of the low pass filter, are supplied as input values, and which provides, at its output, a value representing the difference between the output of the audio level meter and the true signal power of the input signal.

Fixed correction terms for detectors using input filters having fixedattack/release times or other detector types are commonly used in ACvoltage measurement equipment like Multimeters or audio measurementsystems like “Audio Precision”.

Various types of audio level meters exist, like e.g., the VU (VolumeUnit) Meter and the Peak Program Meter, whose readings differ for thesame input signal and are not easy to compare with each other. This isdue to the fact that an audio level meter shall serve two conflictingpurposes. On the one hand, it shall indicate the perceived loudness ofan audio signal, which is related to the signal power. On the otherhand, it shall also indicate the headroom that is still left before thesystem goes into saturation as this would cause audible distortions. Inthis regard, further distinguishing is needed between an analog and adigital system. An analog system overloads relatively gradually whereasa digital one overloads abruptly. Therefore, an object of the inventionis to correct the output level so that it displays the true signalpower, whatever the internal settings of the audio level meter, for asinusoidal signal being supplied to the input.

SUMMARY OF THE INVENTION

The invention is directed to an apparatus and a method for determiningthe difference between the true signal power and an audio level meterreading for various types of audio level meters, including audio levelmeters that, at their inputs, produce the square of the input signal orthe absolute value of the input signal, as presented in claims 1 and 6.Further advantageous embodiments and developments of the invention arepresented in the dependent claims.

BRIEF DESCRIPTION OF THE DRAWINGS

In the following the invention will be described with reference to thedrawings, in which

FIG. 1 shows a block diagram of an audio level meter in accordance withthe invention;

FIG. 2 exemplarily shows the error in dB between the calculatedcorrection value and the correction value obtained from look-up-tables;

FIG. 3 shows exemplary output values of various types of audio levelmeters resulting from the same input signal;

FIG. 4 shows the correction value for an audio level meter using anabsolute peak value as an input signal as a function between the attackand release times;

FIG. 5 shows the correction value for the error shown in FIG. 2 as afunction of the ratio between the attack and release times; and

FIG. 6 shows the error between the calculated correction value and thevalues obtained from look-up-tables.

DETAILED DESCRIPTION OF THE INVENTION

The invention is based upon the finding that, for a sinusoidal inputsignal, the difference between the true signal power and the reading ofan audio level meter substantially only depends from three variables:The treatment of the input signal, i.e. determining the absolute valueor squaring the input signal, the attack time of a lowpass filter usedin the audio level meter and the release time of that filter.

In a corresponding equation phi_0 is the phase angle, for which therising and decaying portions of a signal under test are equal.

In the case of a squared input signal, phi_0 is determined according tothe following equation:

(b−a)*sin(2*phi_(—)0)−2*(b−a)*phi_(—)0*cos(2*phi_(—)0)−a*pi*cos(2*phi_(—)0)=0

In the case of the absolute value of the input signal being determined,phi_0 is determined according to the following equation:

(a−b)*phi_(—)0*sin(phi_(—)0)+(a−b)*cos(phi_(—)0)+b−a*pi/2*sin(phi_(—)0)=0

wherein a=1−exp(−1/(τ_(a)·f_(s))), b=1−exp(−1/(τ_(r)·f_(s))) and f_(s)is the sampling frequency.

When dividing the equations by ‘a’ it will become obvious that phi_0depends merely on the ratio b/a. If τ_(a)>>1/fs and τr>>1/fs b/a issubstantially equal to τ_(a)/τ_(r).

The output signal is then calculated as

output_level=A*sin(phi_(—)0)

wherein A corresponds to the amplitude of the input signal.

The desired reading should, however, correspond to

output_level_(—sin=) A*sin(pi/4).

This results in a difference, expressed in dB, as follows:

Delta_dB=20*log 10(output_level/output_level_sin), or

Delta_dB=20*log 10(sin(phi_(—)0))+3. Note: sin(pi/4)̂2=0.5)

As indicated in the block diagram in FIG. 1, an audio level meteraccording to the invention consists of three building blocks: First, theinput signal x(k) is either squared or rectified, which yields x′(k).Second, a 1st order low-pass filter is applied. Its attack time τ_(a)and release time τ_(r) may differ so that, depending on the currentinput sample x′(k) being greater or smaller than the last output signaly(k−1), two different differential equations are used to calculate thenew output y(k), i.e.,

${y(k)} = \left\{ {\begin{matrix}{{a \cdot {x^{\prime}(k)}} + {\left( {1 - a} \right) \cdot {y\left( {k - 1} \right)}}} \\{{b \cdot {x^{\prime}(k)}} + {\left( {1 - b} \right) \cdot {y\left( {k - 1} \right)}}}\end{matrix},{{if}\mspace{14mu} \begin{matrix}{{x^{\prime}(k)} > {y\left( {k - 1} \right)}} \\{{x^{\prime}(k)} \leq {y\left( {k - 1} \right)}}\end{matrix}}} \right.$

where a=1−exp(−1/(τ_(a)·f_(s))), b=1−exp(−1/(τ_(r)·f_(s))) and f_(s) isthe sampling frequency. In other words, the two differential equationsrepresent a rising or a falling input signal, respectively. Finally, theresult is converted from linear scale to logarithmic scale in decibelsby correspondingly applying a logarithm, i.e., y_(dB)(k)=20log₁₀y(k) ory_(dB)(k)=10log₁₀y(k) depending on whether the input signal wasrectified or squared in the first step. Therefore, there are in totalthree internal parameters, which influence the output of an audio levelmeter: rectification/squaring, attack time τ_(a) and release time τ_(r).Note that an exception to this signal flow is represented by audio levelmeters that shall solely indicate the maximum of the signal to preventany overload from occurring. They either hold a maximum for a presettime or they let it decrease exponentially. FIG. 3 shows exemplaryoutput values of various types of audio level meters resulting from thesame input signal.

Sinusoidal input signals are commonly used as test signals. Especiallyfor non-professional users, it is likely to be rather distracting if thereading of the audio level meter does not match the applied signalpower. As sinusoids are, however, completely defined by their amplitudeand frequency (and, strictly speaking, also their phase), the output ofan audio level meter can be calculated relatively easy for the steadystate (at least approximately and as long as the attack and releasetimes are sufficiently large compared with the input period). Thus, theoutput can be corrected to reflect the true signal power in this case.The analysis results in nonlinear equations, which need to be solvednumerically. To avoid that such a complex task needs to be performedonline, look-up tables with linear interpolation between the entries areused instead. Fortunately, the output level is independent of thefrequency of the input sinusoid. Furthermore, it only depends on theratio between the release and the attack time and not on theirindividual values. Finally, the level offset, or difference, in dB isalso independent of the amplitude of the input signal. Therefore, twoequations, one for a rectified and one for a squared signal, aresufficient to calculate the offset that needs to be applied. Using onlythe indicated sampling points with linear interpolation in between, themaximal error remains below approximately 0.1 dB for the complete rangeof time ratios. Note that an alternative implementation could consist inreplacing the additive correction downstream of the logarithm stage by acorresponding multiplicative one upstream thereof.

FIG. 2 shows the error in dB between the correction value calculatedaccording to the differential equations and the interpolated correctionvalues determined from the look-up table for the correction of an audiolevel meter using a squared audio signal as an input signal. FIG. 5accordingly shows the correction value as a function of the ratiobetween the attack and release times.

FIG. 4 shows the correction value for an audio level meter using anabsolute peak value of the input signal as an input as a function of theratio between the attack and release times. FIG. 6 shows the errorbetween the correction value calculated using the differential equationsand using the look-up table and interpolation.

For testing, a waveform generator producing a sinusoidal test signalsimply needs to be attached to the audio level meter, and the settingsof the waveform generator and the output of the audio level metercompared with each other for τ_(a)≠τ_(r).

1. A method of correcting the output of an audio level meter includingthe steps of: processing an input signal to generate the square or theabsolute value of the input signal; filtering the processed input signalin a low pass filter having a predetermined attack time and releasetime; and converting the output signal from a linear scale to alogarithmic scale; wherein the method further comprises the steps of:calculating a value representing the difference between the output ofthe audio level meter and the true signal power in accordance withwhether the input signal to the audio level meter was subject tosquaring or converting into an absolute value, and in accordance withthe attack and release time of the low pass filter.
 2. The method ofclaim 1, further comprising the step of adding the calculated valuerepresenting the difference between the output of the audio level meterand the true signal power to the output value of the audio level meter.3. The method of claim 1, further comprising the step of multiplying theoutput of the low pass filter with a correction value determined fromthe calculated value representing the difference between the output ofthe audio level meter and the signal power of the input signal prior toconverting the value into logarithmic scale.
 4. The method of claim 1,wherein calculating the value representing the difference between theoutput of the audio level meter and the true signal power is based upona look-up table holding values representing the difference between theoutput of the audio level meter and the signal power of the input signalfor a number of ratios of attack and release time of the low pass filterand for input means forming the absolute value of an input signal and/orinput means squaring the input signal.
 5. The method of claim 4, furthercomprising the step of interpolating values for those ratios notincluded in the look-up table.
 6. A non-transitory computer-readablemedium having computer readable instructions stored thereon forexecution by a processor in an apparatus having an audio level meter toperform a method of correction of the output of the audio level metercomprising: processing an input signal to generate the square or theabsolute value of the input signal; filtering the processed input signalin a low pass filter having a predetermined attack time and releasetime; and converting the output signal from a linear scale to alogarithmic scale; wherein the method further comprises the steps of:calculating a value representing the difference between the output ofthe audio level meter and the true signal power in accordance withwhether the input signal to the audio level meter was subject tosquaring or converting into an absolute value, and in accordance withthe attack and release time of the low pass filter.
 7. Thenon-transitory computer-readable medium of claim 6, further comprisinginstructions for adding the calculated value representing the differencebetween the output of the audio level meter and the true signal power tothe output value of the audio level meter.
 8. The non-transitorycomputer-readable medium of claim 6, further comprising instructions formultiplying the output of the low pass filter with a correction valuedetermined from the calculated value representing the difference betweenthe output of the audio level meter and the signal power of the inputsignal prior to converting the value into logarithmic scale.
 9. Thenon-transitory computer-readable medium of claim 6, wherein calculatingthe value representing the difference between the output of the audiolevel meter and the true signal power is based upon a look-up tableholding values representing the difference between the output of theaudio level meter and the signal power of the input signal for a numberof ratios of attack and release time of the low pass filter and forinput means forming the absolute value of an input signal and/or inputmeans squaring the input signal.
 10. The non-transitorycomputer-readable medium of claim 9, further comprising instructions forinterpolating values for those ratios not included in the look-up table.